Ensure we have (or should have) at least one matching codec before attempting early...

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@58240 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index d33b024264f3ce93ce72f14d710f4f9c08ba3bb5..2ac487f65e50a7feeebeaad49958e3d11b3f387d 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -17056,7 +17056,7 @@ static int sip_sipredirect(struct sip_pvt *p, const char *dest)
 static int sip_get_codec(struct ast_channel *chan)
 {
        struct sip_pvt *p = chan->tech_pvt;
-       return p->peercapability;       
+       return p->peercapability ? p->peercapability : p->capability;   
 }
 
 /*! \brief Send a poke to all known peers 

diff --git a/main/rtp.c b/main/rtp.c
index e25409e47bc6b30153717db94e62154eeb9f9d14..fdeca117c58529d5edc9de361f256f6c54d5451d 100644 (file)
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -1445,7 +1445,7 @@ int ast_rtp_early_bridge(struct ast_channel *dest, struct ast_channel *src)
        struct ast_rtp_protocol *destpr = NULL, *srcpr = NULL;
        enum ast_rtp_get_result audio_dest_res = AST_RTP_GET_FAILED, video_dest_res = AST_RTP_GET_FAILED;
        enum ast_rtp_get_result audio_src_res = AST_RTP_GET_FAILED, video_src_res = AST_RTP_GET_FAILED;
-       int srccodec, nat_active = 0;
+       int srccodec, destcodec, nat_active = 0;
 
        /* Lock channels */
        ast_channel_lock(dest);
@@ -1498,6 +1498,17 @@ int ast_rtp_early_bridge(struct ast_channel *dest, struct ast_channel *src)
                srccodec = srcpr->get_codec(src);
        else
                srccodec = 0;
+       if (audio_dest_res == AST_RTP_TRY_NATIVE && destpr->get_codec)
+               destcodec = destpr->get_codec(dest);
+       else
+               destcodec = 0;
+       /* Ensure we have at least one matching codec */
+       if (!(srccodec & destcodec)) {
+               ast_channel_unlock(dest);
+               if (src)
+                       ast_channel_unlock(src);
+               return 0;
+       }
        /* Consider empty media as non-existant */
        if (audio_src_res == AST_RTP_TRY_NATIVE && !srcp->them.sin_addr.s_addr)
                srcp = NULL;